From 8ad7b84bcaa43f5a4ffe9d8edf6036e4df41859e Mon Sep 17 00:00:00 2001 From: Silvan Calarco Date: Sat, 6 Jan 2024 03:49:58 +0100 Subject: [PATCH] rebuilt with debug package, -tools subpackage and patches from arch linux [release 0.3.6-2mamba;Sat Dec 12 2020] --- ...ail-when-error-occurs-in-parseFormat.patch | 36 ++ ...ays-check-the-number-of-coefficients.patch | 30 ++ libaudiofile-0.3.6-CVE-2015-7747.patch | 156 +++++++ libaudiofile-0.3.6-CVE-2018-13440.patch | 28 ++ libaudiofile-0.3.6-CVE-2018-17095.patch | 26 ++ ...vision-by-zero-in-BlockCodec-runPull.patch | 21 + ...cation-overflow-in-MSADPCM-decodeSam.patch | 116 ++++++ ...multiplication-overflow-in-sfconvert.patch | 66 +++ ...ultiplyCheckOverflow.-It-returns-a-b.patch | 35 ++ ...ues-to-fix-index-overflow-in-IMA.cpp.patch | 33 ++ libaudiofile-0.3.6-gcc6.patch | 102 +++++ libaudiofile-0.3.6-hurd.patch | 381 ++++++++++++++++++ libaudiofile.spec | 74 +++- 13 files changed, 1087 insertions(+), 17 deletions(-) create mode 100644 libaudiofile-0.3.6-Actually-fail-when-error-occurs-in-parseFormat.patch create mode 100644 libaudiofile-0.3.6-Always-check-the-number-of-coefficients.patch create mode 100644 libaudiofile-0.3.6-CVE-2015-7747.patch create mode 100644 libaudiofile-0.3.6-CVE-2018-13440.patch create mode 100644 libaudiofile-0.3.6-CVE-2018-17095.patch create mode 100644 libaudiofile-0.3.6-Check-for-division-by-zero-in-BlockCodec-runPull.patch create mode 100644 libaudiofile-0.3.6-Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch create mode 100644 libaudiofile-0.3.6-Check-for-multiplication-overflow-in-sfconvert.patch create mode 100644 libaudiofile-0.3.6-Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch create mode 100644 libaudiofile-0.3.6-clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch create mode 100644 libaudiofile-0.3.6-gcc6.patch create mode 100644 libaudiofile-0.3.6-hurd.patch diff --git a/libaudiofile-0.3.6-Actually-fail-when-error-occurs-in-parseFormat.patch b/libaudiofile-0.3.6-Actually-fail-when-error-occurs-in-parseFormat.patch new file mode 100644 index 0000000..50cd3dc --- /dev/null +++ b/libaudiofile-0.3.6-Actually-fail-when-error-occurs-in-parseFormat.patch @@ -0,0 +1,36 @@ +From: Antonio Larrosa +Date: Mon, 6 Mar 2017 18:59:26 +0100 +Subject: Actually fail when error occurs in parseFormat + +When there's an unsupported number of bits per sample or an invalid +number of samples per block, don't only print an error message using +the error handler, but actually stop parsing the file. + +This fixes #35 (also reported at +https://bugzilla.opensuse.org/show_bug.cgi?id=1026983 and +https://blogs.gentoo.org/ago/2017/02/20/audiofile-heap-based-buffer-overflow-in-imadecodeblockwave-ima-cpp/ +) +--- + libaudiofile/WAVE.cpp | 2 ++ + 1 file changed, 2 insertions(+) + +diff --git a/libaudiofile/WAVE.cpp b/libaudiofile/WAVE.cpp +index 0fc48e8..d04b796 100644 +--- a/libaudiofile/WAVE.cpp ++++ b/libaudiofile/WAVE.cpp +@@ -332,6 +332,7 @@ status WAVEFile::parseFormat(const Tag &id, uint32_t size) + { + _af_error(AF_BAD_NOT_IMPLEMENTED, + "IMA ADPCM compression supports only 4 bits per sample"); ++ return AF_FAIL; + } + + int bytesPerBlock = (samplesPerBlock + 14) / 8 * 4 * channelCount; +@@ -339,6 +340,7 @@ status WAVEFile::parseFormat(const Tag &id, uint32_t size) + { + _af_error(AF_BAD_CODEC_CONFIG, + "Invalid samples per block for IMA ADPCM compression"); ++ return AF_FAIL; + } + + track->f.sampleWidth = 16; diff --git a/libaudiofile-0.3.6-Always-check-the-number-of-coefficients.patch b/libaudiofile-0.3.6-Always-check-the-number-of-coefficients.patch new file mode 100644 index 0000000..f9427cb --- /dev/null +++ b/libaudiofile-0.3.6-Always-check-the-number-of-coefficients.patch @@ -0,0 +1,30 @@ +From: Antonio Larrosa +Date: Mon, 6 Mar 2017 12:51:22 +0100 +Subject: Always check the number of coefficients + +When building the library with NDEBUG, asserts are eliminated +so it's better to always check that the number of coefficients +is inside the array range. + +This fixes the 00191-audiofile-indexoob issue in #41 +--- + libaudiofile/WAVE.cpp | 6 ++++++ + 1 file changed, 6 insertions(+) + +diff --git a/libaudiofile/WAVE.cpp b/libaudiofile/WAVE.cpp +index 9dd8511..0fc48e8 100644 +--- a/libaudiofile/WAVE.cpp ++++ b/libaudiofile/WAVE.cpp +@@ -281,6 +281,12 @@ status WAVEFile::parseFormat(const Tag &id, uint32_t size) + + /* numCoefficients should be at least 7. */ + assert(numCoefficients >= 7 && numCoefficients <= 255); ++ if (numCoefficients < 7 || numCoefficients > 255) ++ { ++ _af_error(AF_BAD_HEADER, ++ "Bad number of coefficients"); ++ return AF_FAIL; ++ } + + m_msadpcmNumCoefficients = numCoefficients; + diff --git a/libaudiofile-0.3.6-CVE-2015-7747.patch b/libaudiofile-0.3.6-CVE-2015-7747.patch new file mode 100644 index 0000000..3325639 --- /dev/null +++ b/libaudiofile-0.3.6-CVE-2015-7747.patch @@ -0,0 +1,156 @@ +Description: fix buffer overflow when changing both sample format and + number of channels +Origin: https://github.com/mpruett/audiofile/pull/25 +Bug-Ubuntu: https://bugs.launchpad.net/ubuntu/+source/audiofile/+bug/1502721 +Bug-Debian: https://bugs.debian.org/801102 + +--- a/libaudiofile/modules/ModuleState.cpp ++++ b/libaudiofile/modules/ModuleState.cpp +@@ -402,7 +402,7 @@ status ModuleState::arrange(AFfilehandle + addModule(new Transform(outfc, in.pcm, out.pcm)); + + if (in.channelCount != out.channelCount) +- addModule(new ApplyChannelMatrix(infc, isReading, ++ addModule(new ApplyChannelMatrix(outfc, isReading, + in.channelCount, out.channelCount, + in.pcm.minClip, in.pcm.maxClip, + track->channelMatrix)); +--- a/test/Makefile.am ++++ b/test/Makefile.am +@@ -26,6 +26,7 @@ TESTS = \ + VirtualFile \ + floatto24 \ + query2 \ ++ sixteen-stereo-to-eight-mono \ + sixteen-to-eight \ + testchannelmatrix \ + testdouble \ +@@ -139,6 +140,7 @@ printmarkers_SOURCES = printmarkers.c + printmarkers_LDADD = $(LIBAUDIOFILE) -lm + + sixteen_to_eight_SOURCES = sixteen-to-eight.c TestUtilities.cpp TestUtilities.h ++sixteen_stereo_to_eight_mono_SOURCES = sixteen-stereo-to-eight-mono.c TestUtilities.cpp TestUtilities.h + + testchannelmatrix_SOURCES = testchannelmatrix.c TestUtilities.cpp TestUtilities.h + +--- /dev/null ++++ b/test/sixteen-stereo-to-eight-mono.c +@@ -0,0 +1,118 @@ ++/* ++ Audio File Library ++ ++ Copyright 2000, Silicon Graphics, Inc. ++ ++ This program is free software; you can redistribute it and/or modify ++ it under the terms of the GNU General Public License as published by ++ the Free Software Foundation; either version 2 of the License, or ++ (at your option) any later version. ++ ++ This program is distributed in the hope that it will be useful, ++ but WITHOUT ANY WARRANTY; without even the implied warranty of ++ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the ++ GNU General Public License for more details. ++ ++ You should have received a copy of the GNU General Public License along ++ with this program; if not, write to the Free Software Foundation, Inc., ++ 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. ++*/ ++ ++/* ++ sixteen-stereo-to-eight-mono.c ++ ++ This program tests the conversion from 2-channel 16-bit integers to ++ 1-channel 8-bit integers. ++*/ ++ ++#ifdef HAVE_CONFIG_H ++#include ++#endif ++ ++#include ++#include ++#include ++#include ++#include ++#include ++ ++#include ++ ++#include "TestUtilities.h" ++ ++int main (int argc, char **argv) ++{ ++ AFfilehandle file; ++ AFfilesetup setup; ++ int16_t frames16[] = {14298, 392, 3923, -683, 958, -1921}; ++ int8_t frames8[] = {28, 6, -2}; ++ int i, frameCount = 3; ++ int8_t byte; ++ AFframecount result; ++ ++ setup = afNewFileSetup(); ++ ++ afInitFileFormat(setup, AF_FILE_WAVE); ++ ++ afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16); ++ afInitChannels(setup, AF_DEFAULT_TRACK, 2); ++ ++ char *testFileName; ++ if (!createTemporaryFile("sixteen-to-eight", &testFileName)) ++ { ++ fprintf(stderr, "Could not create temporary file.\n"); ++ exit(EXIT_FAILURE); ++ } ++ ++ file = afOpenFile(testFileName, "w", setup); ++ if (file == AF_NULL_FILEHANDLE) ++ { ++ fprintf(stderr, "could not open file for writing\n"); ++ exit(EXIT_FAILURE); ++ } ++ ++ afFreeFileSetup(setup); ++ ++ afWriteFrames(file, AF_DEFAULT_TRACK, frames16, frameCount); ++ ++ afCloseFile(file); ++ ++ file = afOpenFile(testFileName, "r", AF_NULL_FILESETUP); ++ if (file == AF_NULL_FILEHANDLE) ++ { ++ fprintf(stderr, "could not open file for reading\n"); ++ exit(EXIT_FAILURE); ++ } ++ ++ afSetVirtualSampleFormat(file, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 8); ++ afSetVirtualChannels(file, AF_DEFAULT_TRACK, 1); ++ ++ for (i=0; i +Date: Thu, 27 Sep 2018 10:48:45 +0200 +Subject: [PATCH] ModuleState: handle compress/decompress init failure + +When the unit initcompress or initdecompress function fails, +m_fileModule is NULL. Return AF_FAIL in that case instead of +causing NULL pointer dereferences later. + +Fixes #49 +--- + libaudiofile/modules/ModuleState.cpp | 3 +++ + 1 file changed, 3 insertions(+) + +diff --git a/libaudiofile/modules/ModuleState.cpp b/libaudiofile/modules/ModuleState.cpp +index 0c29d7a..070fd9b 100644 +--- a/libaudiofile/modules/ModuleState.cpp ++++ b/libaudiofile/modules/ModuleState.cpp +@@ -75,6 +75,9 @@ status ModuleState::initFileModule(AFfilehandle file, Track *track) + m_fileModule = unit->initcompress(track, file->m_fh, file->m_seekok, + file->m_fileFormat == AF_FILE_RAWDATA, &chunkFrames); + ++ if (!m_fileModule) ++ return AF_FAIL; ++ + if (unit->needsRebuffer) + { + assert(unit->nativeSampleFormat == AF_SAMPFMT_TWOSCOMP); diff --git a/libaudiofile-0.3.6-CVE-2018-17095.patch b/libaudiofile-0.3.6-CVE-2018-17095.patch new file mode 100644 index 0000000..231021b --- /dev/null +++ b/libaudiofile-0.3.6-CVE-2018-17095.patch @@ -0,0 +1,26 @@ +From 822b732fd31ffcb78f6920001e9b1fbd815fa712 Mon Sep 17 00:00:00 2001 +From: Wim Taymans +Date: Thu, 27 Sep 2018 12:11:12 +0200 +Subject: [PATCH] SimpleModule: set output chunk framecount after pull + +After pulling the data, set the output chunk to the amount of +frames we pulled so that the next module in the chain has the correct +frame count. + +Fixes #50 and #51 +--- + libaudiofile/modules/SimpleModule.cpp | 1 + + 1 file changed, 1 insertion(+) + +diff --git a/libaudiofile/modules/SimpleModule.cpp b/libaudiofile/modules/SimpleModule.cpp +index 2bae1eb..e87932c 100644 +--- a/libaudiofile/modules/SimpleModule.cpp ++++ b/libaudiofile/modules/SimpleModule.cpp +@@ -26,6 +26,7 @@ + void SimpleModule::runPull() + { + pull(m_outChunk->frameCount); ++ m_outChunk->frameCount = m_inChunk->frameCount; + run(*m_inChunk, *m_outChunk); + } + diff --git a/libaudiofile-0.3.6-Check-for-division-by-zero-in-BlockCodec-runPull.patch b/libaudiofile-0.3.6-Check-for-division-by-zero-in-BlockCodec-runPull.patch new file mode 100644 index 0000000..e001133 --- /dev/null +++ b/libaudiofile-0.3.6-Check-for-division-by-zero-in-BlockCodec-runPull.patch @@ -0,0 +1,21 @@ +From: Antonio Larrosa +Date: Thu, 9 Mar 2017 10:21:18 +0100 +Subject: Check for division by zero in BlockCodec::runPull + +--- + libaudiofile/modules/BlockCodec.cpp | 2 +- + 1 file changed, 1 insertion(+), 1 deletion(-) + +diff --git a/libaudiofile/modules/BlockCodec.cpp b/libaudiofile/modules/BlockCodec.cpp +index 4731be1..eb2fb4d 100644 +--- a/libaudiofile/modules/BlockCodec.cpp ++++ b/libaudiofile/modules/BlockCodec.cpp +@@ -47,7 +47,7 @@ void BlockCodec::runPull() + + // Read the compressed data. + ssize_t bytesRead = read(m_inChunk->buffer, m_bytesPerPacket * blockCount); +- int blocksRead = bytesRead >= 0 ? bytesRead / m_bytesPerPacket : 0; ++ int blocksRead = (bytesRead >= 0 && m_bytesPerPacket > 0) ? bytesRead / m_bytesPerPacket : 0; + + // Decompress into m_outChunk. + for (int i=0; i +Date: Mon, 6 Mar 2017 13:43:53 +0100 +Subject: Check for multiplication overflow in MSADPCM decodeSample + +Check for multiplication overflow (using __builtin_mul_overflow +if available) in MSADPCM.cpp decodeSample and return an empty +decoded block if an error occurs. + +This fixes the 00193-audiofile-signintoverflow-MSADPCM case of #41 +--- + libaudiofile/modules/BlockCodec.cpp | 5 ++-- + libaudiofile/modules/MSADPCM.cpp | 47 +++++++++++++++++++++++++++++++++---- + 2 files changed, 46 insertions(+), 6 deletions(-) + +diff --git a/libaudiofile/modules/BlockCodec.cpp b/libaudiofile/modules/BlockCodec.cpp +index 45925e8..4731be1 100644 +--- a/libaudiofile/modules/BlockCodec.cpp ++++ b/libaudiofile/modules/BlockCodec.cpp +@@ -52,8 +52,9 @@ void BlockCodec::runPull() + // Decompress into m_outChunk. + for (int i=0; i(m_inChunk->buffer) + i * m_bytesPerPacket, +- static_cast(m_outChunk->buffer) + i * m_framesPerPacket * m_track->f.channelCount); ++ if (decodeBlock(static_cast(m_inChunk->buffer) + i * m_bytesPerPacket, ++ static_cast(m_outChunk->buffer) + i * m_framesPerPacket * m_track->f.channelCount)==0) ++ break; + + framesRead += m_framesPerPacket; + } +diff --git a/libaudiofile/modules/MSADPCM.cpp b/libaudiofile/modules/MSADPCM.cpp +index 8ea3c85..ef9c38c 100644 +--- a/libaudiofile/modules/MSADPCM.cpp ++++ b/libaudiofile/modules/MSADPCM.cpp +@@ -101,24 +101,60 @@ static const int16_t adaptationTable[] = + 768, 614, 512, 409, 307, 230, 230, 230 + }; + ++int firstBitSet(int x) ++{ ++ int position=0; ++ while (x!=0) ++ { ++ x>>=1; ++ ++position; ++ } ++ return position; ++} ++ ++#ifndef __has_builtin ++#define __has_builtin(x) 0 ++#endif ++ ++int multiplyCheckOverflow(int a, int b, int *result) ++{ ++#if (defined __GNUC__ && __GNUC__ >= 5) || ( __clang__ && __has_builtin(__builtin_mul_overflow)) ++ return __builtin_mul_overflow(a, b, result); ++#else ++ if (firstBitSet(a)+firstBitSet(b)>31) // int is signed, so we can't use 32 bits ++ return true; ++ *result = a * b; ++ return false; ++#endif ++} ++ ++ + // Compute a linear PCM value from the given differential coded value. + static int16_t decodeSample(ms_adpcm_state &state, +- uint8_t code, const int16_t *coefficient) ++ uint8_t code, const int16_t *coefficient, bool *ok=NULL) + { + int linearSample = (state.sample1 * coefficient[0] + + state.sample2 * coefficient[1]) >> 8; ++ int delta; + + linearSample += ((code & 0x08) ? (code - 0x10) : code) * state.delta; + + linearSample = clamp(linearSample, MIN_INT16, MAX_INT16); + +- int delta = (state.delta * adaptationTable[code]) >> 8; ++ if (multiplyCheckOverflow(state.delta, adaptationTable[code], &delta)) ++ { ++ if (ok) *ok=false; ++ _af_error(AF_BAD_COMPRESSION, "Error decoding sample"); ++ return 0; ++ } ++ delta >>= 8; + if (delta < 16) + delta = 16; + + state.delta = delta; + state.sample2 = state.sample1; + state.sample1 = linearSample; ++ if (ok) *ok=true; + + return static_cast(linearSample); + } +@@ -212,13 +248,16 @@ int MSADPCM::decodeBlock(const uint8_t *encoded, int16_t *decoded) + { + uint8_t code; + int16_t newSample; ++ bool ok; + + code = *encoded >> 4; +- newSample = decodeSample(*state[0], code, coefficient[0]); ++ newSample = decodeSample(*state[0], code, coefficient[0], &ok); ++ if (!ok) return 0; + *decoded++ = newSample; + + code = *encoded & 0x0f; +- newSample = decodeSample(*state[1], code, coefficient[1]); ++ newSample = decodeSample(*state[1], code, coefficient[1], &ok); ++ if (!ok) return 0; + *decoded++ = newSample; + + encoded++; diff --git a/libaudiofile-0.3.6-Check-for-multiplication-overflow-in-sfconvert.patch b/libaudiofile-0.3.6-Check-for-multiplication-overflow-in-sfconvert.patch new file mode 100644 index 0000000..0f17140 --- /dev/null +++ b/libaudiofile-0.3.6-Check-for-multiplication-overflow-in-sfconvert.patch @@ -0,0 +1,66 @@ +From: Antonio Larrosa +Date: Mon, 6 Mar 2017 13:54:52 +0100 +Subject: Check for multiplication overflow in sfconvert + +Checks that a multiplication doesn't overflow when +calculating the buffer size, and if it overflows, +reduce the buffer size instead of failing. + +This fixes the 00192-audiofile-signintoverflow-sfconvert case +in #41 +--- + sfcommands/sfconvert.c | 34 ++++++++++++++++++++++++++++++++-- + 1 file changed, 32 insertions(+), 2 deletions(-) + +diff --git a/sfcommands/sfconvert.c b/sfcommands/sfconvert.c +index 80a1bc4..970a3e4 100644 +--- a/sfcommands/sfconvert.c ++++ b/sfcommands/sfconvert.c +@@ -45,6 +45,33 @@ void printusage (void); + void usageerror (void); + bool copyaudiodata (AFfilehandle infile, AFfilehandle outfile, int trackid); + ++int firstBitSet(int x) ++{ ++ int position=0; ++ while (x!=0) ++ { ++ x>>=1; ++ ++position; ++ } ++ return position; ++} ++ ++#ifndef __has_builtin ++#define __has_builtin(x) 0 ++#endif ++ ++int multiplyCheckOverflow(int a, int b, int *result) ++{ ++#if (defined __GNUC__ && __GNUC__ >= 5) || ( __clang__ && __has_builtin(__builtin_mul_overflow)) ++ return __builtin_mul_overflow(a, b, result); ++#else ++ if (firstBitSet(a)+firstBitSet(b)>31) // int is signed, so we can't use 32 bits ++ return true; ++ *result = a * b; ++ return false; ++#endif ++} ++ + int main (int argc, char **argv) + { + if (argc == 2) +@@ -323,8 +350,11 @@ bool copyaudiodata (AFfilehandle infile, AFfilehandle outfile, int trackid) + { + int frameSize = afGetVirtualFrameSize(infile, trackid, 1); + +- const int kBufferFrameCount = 65536; +- void *buffer = malloc(kBufferFrameCount * frameSize); ++ int kBufferFrameCount = 65536; ++ int bufferSize; ++ while (multiplyCheckOverflow(kBufferFrameCount, frameSize, &bufferSize)) ++ kBufferFrameCount /= 2; ++ void *buffer = malloc(bufferSize); + + AFframecount totalFrames = afGetFrameCount(infile, AF_DEFAULT_TRACK); + AFframecount totalFramesWritten = 0; diff --git a/libaudiofile-0.3.6-Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch b/libaudiofile-0.3.6-Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch new file mode 100644 index 0000000..35627d3 --- /dev/null +++ b/libaudiofile-0.3.6-Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch @@ -0,0 +1,35 @@ +From: Antonio Larrosa +Date: Fri, 10 Mar 2017 15:40:02 +0100 +Subject: Fix signature of multiplyCheckOverflow. It returns a bool, not an int + +--- + libaudiofile/modules/MSADPCM.cpp | 2 +- + sfcommands/sfconvert.c | 2 +- + 2 files changed, 2 insertions(+), 2 deletions(-) + +diff --git a/libaudiofile/modules/MSADPCM.cpp b/libaudiofile/modules/MSADPCM.cpp +index ef9c38c..d8c9553 100644 +--- a/libaudiofile/modules/MSADPCM.cpp ++++ b/libaudiofile/modules/MSADPCM.cpp +@@ -116,7 +116,7 @@ int firstBitSet(int x) + #define __has_builtin(x) 0 + #endif + +-int multiplyCheckOverflow(int a, int b, int *result) ++bool multiplyCheckOverflow(int a, int b, int *result) + { + #if (defined __GNUC__ && __GNUC__ >= 5) || ( __clang__ && __has_builtin(__builtin_mul_overflow)) + return __builtin_mul_overflow(a, b, result); +diff --git a/sfcommands/sfconvert.c b/sfcommands/sfconvert.c +index 970a3e4..367f7a5 100644 +--- a/sfcommands/sfconvert.c ++++ b/sfcommands/sfconvert.c +@@ -60,7 +60,7 @@ int firstBitSet(int x) + #define __has_builtin(x) 0 + #endif + +-int multiplyCheckOverflow(int a, int b, int *result) ++bool multiplyCheckOverflow(int a, int b, int *result) + { + #if (defined __GNUC__ && __GNUC__ >= 5) || ( __clang__ && __has_builtin(__builtin_mul_overflow)) + return __builtin_mul_overflow(a, b, result); diff --git a/libaudiofile-0.3.6-clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch b/libaudiofile-0.3.6-clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch new file mode 100644 index 0000000..c1047af --- /dev/null +++ b/libaudiofile-0.3.6-clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch @@ -0,0 +1,33 @@ +From: Antonio Larrosa +Date: Mon, 6 Mar 2017 18:02:31 +0100 +Subject: clamp index values to fix index overflow in IMA.cpp + +This fixes #33 +(also reported at https://bugzilla.opensuse.org/show_bug.cgi?id=1026981 +and https://blogs.gentoo.org/ago/2017/02/20/audiofile-global-buffer-overflow-in-decodesample-ima-cpp/) +--- + libaudiofile/modules/IMA.cpp | 4 ++-- + 1 file changed, 2 insertions(+), 2 deletions(-) + +diff --git a/libaudiofile/modules/IMA.cpp b/libaudiofile/modules/IMA.cpp +index 7476d44..df4aad6 100644 +--- a/libaudiofile/modules/IMA.cpp ++++ b/libaudiofile/modules/IMA.cpp +@@ -169,7 +169,7 @@ int IMA::decodeBlockWAVE(const uint8_t *encoded, int16_t *decoded) + if (encoded[1] & 0x80) + m_adpcmState[c].previousValue -= 0x10000; + +- m_adpcmState[c].index = encoded[2]; ++ m_adpcmState[c].index = clamp(encoded[2], 0, 88); + + *decoded++ = m_adpcmState[c].previousValue; + +@@ -210,7 +210,7 @@ int IMA::decodeBlockQT(const uint8_t *encoded, int16_t *decoded) + predictor -= 0x10000; + + state.previousValue = clamp(predictor, MIN_INT16, MAX_INT16); +- state.index = encoded[1] & 0x7f; ++ state.index = clamp(encoded[1] & 0x7f, 0, 88); + encoded += 2; + + for (int n=0; n +Origin: vendor, https://github.com/mpruett/audiofile/pull/27 +Bug-Debian: https://bugs.debian.org/812055 +--- +This patch header follows DEP-3: http://dep.debian.net/deps/dep3/ + +--- a/libaudiofile/modules/SimpleModule.h ++++ b/libaudiofile/modules/SimpleModule.h +@@ -123,7 +123,7 @@ struct signConverter + typedef typename IntTypes::UnsignedType UnsignedType; + + static const int kScaleBits = (Format + 1) * CHAR_BIT - 1; +- static const int kMinSignedValue = -1 << kScaleBits; ++ static const int kMinSignedValue = 0-(1U< + { +--- a/test/FloatToInt.cpp ++++ b/test/FloatToInt.cpp +@@ -115,7 +115,7 @@ TEST_F(FloatToIntTest, Int16) + EXPECT_EQ(readData[i], expectedData[i]); + } + +-static const int32_t kMinInt24 = -1<<23; ++static const int32_t kMinInt24 = 0-(1U<<23); + static const int32_t kMaxInt24 = (1<<23) - 1; + + TEST_F(FloatToIntTest, Int24) +--- a/test/IntToFloat.cpp ++++ b/test/IntToFloat.cpp +@@ -117,7 +117,7 @@ TEST_F(IntToFloatTest, Int16) + EXPECT_EQ(readData[i], expectedData[i]); + } + +-static const int32_t kMinInt24 = -1<<23; ++static const int32_t kMinInt24 = 0-(1U<<23); + static const int32_t kMaxInt24 = (1<<23) - 1; + + TEST_F(IntToFloatTest, Int24) +--- a/test/NeXT.cpp ++++ b/test/NeXT.cpp +@@ -37,13 +37,13 @@ + + #include "TestUtilities.h" + +-const char kDataUnspecifiedLength[] = ++const signed char kDataUnspecifiedLength[] = + { + '.', 's', 'n', 'd', + 0, 0, 0, 24, // offset of 24 bytes +- 0xff, 0xff, 0xff, 0xff, // unspecified length ++ -1, -1, -1, -1, // unspecified length + 0, 0, 0, 3, // 16-bit linear +- 0, 0, 172, 68, // 44100 Hz ++ 0, 0, -84, 68, // 44100 Hz (0xAC44) + 0, 0, 0, 1, // 1 channel + 0, 1, + 0, 1, +@@ -57,13 +57,13 @@ const char kDataUnspecifiedLength[] = + 0, 55 + }; + +-const char kDataTruncated[] = ++const signed char kDataTruncated[] = + { + '.', 's', 'n', 'd', + 0, 0, 0, 24, // offset of 24 bytes + 0, 0, 0, 20, // length of 20 bytes + 0, 0, 0, 3, // 16-bit linear +- 0, 0, 172, 68, // 44100 Hz ++ 0, 0, -84, 68, // 44100 Hz (0xAC44) + 0, 0, 0, 1, // 1 channel + 0, 1, + 0, 1, +@@ -152,13 +152,13 @@ TEST(NeXT, Truncated) + ASSERT_EQ(::unlink(testFileName.c_str()), 0); + } + +-const char kDataZeroChannels[] = ++const signed char kDataZeroChannels[] = + { + '.', 's', 'n', 'd', + 0, 0, 0, 24, // offset of 24 bytes + 0, 0, 0, 2, // 2 bytes + 0, 0, 0, 3, // 16-bit linear +- 0, 0, 172, 68, // 44100 Hz ++ 0, 0, -84, 68, // 44100 Hz (0xAC44) + 0, 0, 0, 0, // 0 channels + 0, 1 + }; +--- a/test/Sign.cpp ++++ b/test/Sign.cpp +@@ -116,7 +116,7 @@ TEST_F(SignConversionTest, Int16) + EXPECT_EQ(readData[i], expectedData[i]); + } + +-static const int32_t kMinInt24 = -1<<23; ++static const int32_t kMinInt24 = 0-(1U<<23); + static const int32_t kMaxInt24 = (1<<23) - 1; + static const uint32_t kMaxUInt24 = (1<<24) - 1; + diff --git a/libaudiofile-0.3.6-hurd.patch b/libaudiofile-0.3.6-hurd.patch new file mode 100644 index 0000000..b5941dc --- /dev/null +++ b/libaudiofile-0.3.6-hurd.patch @@ -0,0 +1,381 @@ +Description: Remove usage of PATH_MAX in tests to fix FTBFS on Hurd. + jcowgill: Removed Changelog changes +Author: Pino Toscano +Origin: backport, https://github.com/mpruett/audiofile/commit/34c261034f1193a783196618f0052112e00fbcfe +Bug: https://github.com/mpruett/audiofile/pull/17 +Bug-Debian: https://bugs.debian.org/762595 +--- +This patch header follows DEP-3: http://dep.debian.net/deps/dep3/ + +--- a/test/TestUtilities.cpp ++++ b/test/TestUtilities.cpp +@@ -21,8 +21,8 @@ + #include "TestUtilities.h" + + #include +-#include + #include ++#include + #include + + bool createTemporaryFile(const std::string &prefix, std::string *path) +@@ -35,12 +35,12 @@ bool createTemporaryFile(const std::stri + return true; + } + +-bool createTemporaryFile(const char *prefix, char *path) ++bool createTemporaryFile(const char *prefix, char **path) + { +- snprintf(path, PATH_MAX, "/tmp/%s-XXXXXX", prefix); +- int fd = ::mkstemp(path); +- if (fd < 0) +- return false; +- ::close(fd); +- return true; ++ *path = NULL; ++ std::string pathString; ++ bool result = createTemporaryFile(prefix, &pathString); ++ if (result) ++ *path = ::strdup(pathString.c_str()); ++ return result; + } +--- a/test/TestUtilities.h ++++ b/test/TestUtilities.h +@@ -53,7 +53,7 @@ extern "C" { + + #include + +-bool createTemporaryFile(const char *prefix, char *path); ++bool createTemporaryFile(const char *prefix, char **path); + + #ifdef __cplusplus + } +--- a/test/floatto24.c ++++ b/test/floatto24.c +@@ -86,8 +86,8 @@ int main (int argc, char **argv) + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_FLOAT, 32); + +- char testFileName[PATH_MAX]; +- if (!createTemporaryFile("floatto24", testFileName)) ++ char *testFileName; ++ if (!createTemporaryFile("floatto24", &testFileName)) + { + fprintf(stderr, "Could not create temporary file.\n"); + exit(EXIT_FAILURE); +@@ -182,6 +182,7 @@ int main (int argc, char **argv) + } + + unlink(testFileName); ++ free(testFileName); + + exit(EXIT_SUCCESS); + } +--- a/test/sixteen-to-eight.c ++++ b/test/sixteen-to-eight.c +@@ -57,8 +57,8 @@ int main (int argc, char **argv) + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_UNSIGNED, 8); + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + +- char testFileName[PATH_MAX]; +- if (!createTemporaryFile("sixteen-to-eight", testFileName)) ++ char *testFileName; ++ if (!createTemporaryFile("sixteen-to-eight", &testFileName)) + { + fprintf(stderr, "Could not create temporary file.\n"); + exit(EXIT_FAILURE); +@@ -113,6 +113,7 @@ int main (int argc, char **argv) + + afCloseFile(file); + unlink(testFileName); ++ free(testFileName); + + exit(EXIT_SUCCESS); + } +--- a/test/testchannelmatrix.c ++++ b/test/testchannelmatrix.c +@@ -39,7 +39,7 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + const short samples[] = {300, -300, 515, -515, 2315, -2315, 9154, -9154}; + #define SAMPLE_COUNT (sizeof (samples) / sizeof (short)) +@@ -47,7 +47,11 @@ const short samples[] = {300, -300, 515, + + void cleanup (void) + { +- unlink(sTestFileName); ++ if (sTestFileName) ++ { ++ unlink(sTestFileName); ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -76,7 +80,7 @@ int main (void) + afInitFileFormat(setup, AF_FILE_AIFFC); + + /* Write stereo data to test file. */ +- ensure(createTemporaryFile("testchannelmatrix", sTestFileName), ++ ensure(createTemporaryFile("testchannelmatrix", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + ensure(file != AF_NULL_FILEHANDLE, "could not open file for writing"); +--- a/test/testdouble.c ++++ b/test/testdouble.c +@@ -38,7 +38,7 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + const double samples[] = + {1.0, 0.6, -0.3, 0.95, 0.2, -0.6, 0.9, 0.4, -0.22, 0.125, 0.1, -0.4}; +@@ -48,7 +48,11 @@ void testdouble (int fileFormat); + + void cleanup (void) + { +- unlink(sTestFileName); ++ if (sTestFileName) ++ { ++ unlink(sTestFileName); ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -96,7 +100,7 @@ void testdouble (int fileFormat) + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_DOUBLE, 64); + afInitChannels(setup, AF_DEFAULT_TRACK, 2); + +- ensure(createTemporaryFile("testdouble", sTestFileName), ++ ensure(createTemporaryFile("testdouble", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + ensure(file != AF_NULL_FILEHANDLE, "could not open file for writing"); +--- a/test/testfloat.c ++++ b/test/testfloat.c +@@ -38,7 +38,7 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + const float samples[] = + {1.0, 0.6, -0.3, 0.95, 0.2, -0.6, 0.9, 0.4, -0.22, 0.125, 0.1, -0.4}; +@@ -48,7 +48,11 @@ void testfloat (int fileFormat); + + void cleanup (void) + { +- unlink(sTestFileName); ++ if (sTestFileName) ++ { ++ unlink(sTestFileName); ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -96,7 +100,7 @@ void testfloat (int fileFormat) + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_FLOAT, 32); + afInitChannels(setup, AF_DEFAULT_TRACK, 2); + +- ensure(createTemporaryFile("testfloat", sTestFileName), ++ ensure(createTemporaryFile("testfloat", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + ensure(file != AF_NULL_FILEHANDLE, "could not open file for writing"); +--- a/test/testmarkers.c ++++ b/test/testmarkers.c +@@ -32,15 +32,19 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + #define FRAME_COUNT 200 + + void cleanup (void) + { ++ if (sTestFileName) ++ { + #ifndef DEBUG +- unlink(sTestFileName); ++ unlink(sTestFileName); + #endif ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -127,7 +131,7 @@ int testmarkers (int fileformat) + + int main (void) + { +- ensure(createTemporaryFile("testmarkers", sTestFileName), ++ ensure(createTemporaryFile("testmarkers", &sTestFileName), + "could not create temporary file"); + + testmarkers(AF_FILE_AIFF); +--- a/test/twentyfour.c ++++ b/test/twentyfour.c +@@ -71,8 +71,8 @@ int main (int argc, char **argv) + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 24); + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + +- char testFileName[PATH_MAX]; +- if (!createTemporaryFile("twentyfour", testFileName)) ++ char *testFileName; ++ if (!createTemporaryFile("twentyfour", &testFileName)) + { + fprintf(stderr, "could not create temporary file\n"); + exit(EXIT_FAILURE); +@@ -239,6 +239,7 @@ int main (int argc, char **argv) + exit(EXIT_FAILURE); + } + unlink(testFileName); ++ free(testFileName); + + exit(EXIT_SUCCESS); + } +--- a/test/twentyfour2.c ++++ b/test/twentyfour2.c +@@ -45,15 +45,19 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + #define FRAME_COUNT 10000 + + void cleanup (void) + { ++ if (sTestFileName) ++ { + #ifndef DEBUG +- unlink(sTestFileName); ++ unlink(sTestFileName); + #endif ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -78,7 +82,7 @@ int main (void) + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 24); + +- ensure(createTemporaryFile("twentyfour2", sTestFileName), ++ ensure(createTemporaryFile("twentyfour2", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + ensure(file != NULL, "could not open test file for writing"); +--- a/test/writealaw.c ++++ b/test/writealaw.c +@@ -53,7 +53,7 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + #define FRAME_COUNT 16 + #define SAMPLE_COUNT FRAME_COUNT +@@ -62,9 +62,13 @@ void testalaw (int fileFormat); + + void cleanup (void) + { ++ if (sTestFileName) ++ { + #ifndef DEBUG +- unlink(sTestFileName); ++ unlink(sTestFileName); + #endif ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -113,7 +117,7 @@ void testalaw (int fileFormat) + afInitFileFormat(setup, fileFormat); + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + +- ensure(createTemporaryFile("writealaw", sTestFileName), ++ ensure(createTemporaryFile("writealaw", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + afFreeFileSetup(setup); +--- a/test/writeraw.c ++++ b/test/writeraw.c +@@ -44,13 +44,17 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + void cleanup (void) + { ++ if (sTestFileName) ++ { + #ifndef DEBUG +- unlink(sTestFileName); ++ unlink(sTestFileName); + #endif ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -84,7 +88,7 @@ int main (int argc, char **argv) + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + afInitSampleFormat(setup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16); + +- ensure(createTemporaryFile("writeraw", sTestFileName), ++ ensure(createTemporaryFile("writeraw", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + ensure(file != AF_NULL_FILEHANDLE, "unable to open file for writing"); +--- a/test/writeulaw.c ++++ b/test/writeulaw.c +@@ -53,7 +53,7 @@ + + #include "TestUtilities.h" + +-static char sTestFileName[PATH_MAX]; ++static char *sTestFileName; + + #define FRAME_COUNT 16 + #define SAMPLE_COUNT FRAME_COUNT +@@ -62,9 +62,13 @@ void testulaw (int fileFormat); + + void cleanup (void) + { ++ if (sTestFileName) ++ { + #ifndef DEBUG +- unlink(sTestFileName); ++ unlink(sTestFileName); + #endif ++ free(sTestFileName); ++ } + } + + void ensure (int condition, const char *message) +@@ -113,7 +117,7 @@ void testulaw (int fileFormat) + afInitFileFormat(setup, fileFormat); + afInitChannels(setup, AF_DEFAULT_TRACK, 1); + +- ensure(createTemporaryFile("writeulaw", sTestFileName), ++ ensure(createTemporaryFile("writeulaw", &sTestFileName), + "could not create temporary file"); + file = afOpenFile(sTestFileName, "w", setup); + afFreeFileSetup(setup); diff --git a/libaudiofile.spec b/libaudiofile.spec index 9b7f2c3..400841a 100644 --- a/libaudiofile.spec +++ b/libaudiofile.spec @@ -1,6 +1,6 @@ Name: libaudiofile Version: 0.3.6 -Release: 1mamba +Release: 2mamba Summary: A uniform programming interface to standard digital audio file formats Group: System/Kernel and Hardware Vendor: openmamba @@ -9,9 +9,26 @@ Packager: Silvan Calarco URL: http://www.68k.org/~michael/audiofile Source: http://www.68k.org/~michael/audiofile/audiofile-%{version}.tar.gz Patch: %{name}-0.2.6-m4_underquoted_warning.patch +Patch1: libaudiofile-0.3.6-gcc6.patch +Patch2: libaudiofile-0.3.6-hurd.patch +Patch3: libaudiofile-0.3.6-CVE-2015-7747.patch +Patch4: libaudiofile-0.3.6-clamp-index-values-to-fix-index-overflow-in-IMA.cpp.patch +Patch5: libaudiofile-0.3.6-Always-check-the-number-of-coefficients.patch +Patch6: libaudiofile-0.3.6-Check-for-multiplication-overflow-in-MSADPCM-decodeSam.patch +Patch7: libaudiofile-0.3.6-Check-for-multiplication-overflow-in-sfconvert.patch +Patch8: libaudiofile-0.3.6-Fix-signature-of-multiplyCheckOverflow.-It-returns-a-b.patch +Patch9: libaudiofile-0.3.6-Actually-fail-when-error-occurs-in-parseFormat.patch +Patch10: libaudiofile-0.3.6-Check-for-division-by-zero-in-BlockCodec-runPull.patch +Patch11: libaudiofile-0.3.6-CVE-2018-13440.patch +Patch12: libaudiofile-0.3.6-CVE-2018-17095.patch License: LGPL ## AUTOBUILDREQ-BEGIN BuildRequires: glibc-devel +BuildRequires: ldconfig +BuildRequires: libflac-devel +BuildRequires: libgcc +BuildRequires: libogg-devel +BuildRequires: libstdc++6-devel ## AUTOBUILDREQ-END BuildRoot: %{_tmppath}/%{name}-%{version}-root @@ -29,22 +46,38 @@ Group: Development/Libraries Requires: %{name} = %{version} %description devel -The Silicon Graphics Audio File Library provides a uniform programming interface to standard digital audio file formats. -Currently supported sound file formats include AIFF/AIFF-C, WAVE, NeXT/Sun .snd/.au, and Berkeley/IRCAM/CARL. -Supported compression formats are currently G.711 mu-law and A-law and IMA and MS ADPCM. -Key goals of the Audio File Library are file format transparency and data format transparency. -The same calls for opening a file, accessing and manipulating audio metadata (e.g., sample rate, sample format, textual information, MIDI parameters), and reading/writing sample data will work with any supported audio file format. -Likewise, the format of the audio data presented to the application need not be tied to the format of the data contained in the file. +This package contains static libraries and header files needed for development with %{name}. -This package contains static libraries and header files need for development. +%package tools +Summary: Tools provided with %{name} +Group: Applications/Multimedia +Requires: %{name} = %{version} + +%description tools +This package contains sthe tools provided with %{name}. + +%debug_package %prep %setup -q -n audiofile-%{version} #%patch -p1 - +%patch1 -p1 +%patch2 -p1 +%patch3 -p1 +%patch4 -p1 +%patch5 -p1 +%patch6 -p1 +%patch7 -p1 +%patch8 -p1 +%patch9 -p1 +%patch10 -p1 +%patch11 -p1 +%patch12 -p1 +autoreconf -vfi + %build -%configure \ - --with-kernel=/usr/src/linux-`uname -r` +%configure + %make %install @@ -59,23 +92,30 @@ This package contains static libraries and header files need for development. %files %defattr(-,root,root) -%{_bindir}/* -%{_libdir}/*.so.* -%{_mandir}/man1/* +%{_libdir}/libaudiofile.so.* %doc AUTHORS COPYING %files devel %defattr(-,root,root) -%{_libdir}/*.a -%{_libdir}/*.la -%{_libdir}/*.so +%{_libdir}/libaudiofile.a +%{_libdir}/libaudiofile.la +%{_libdir}/libaudiofile.so %{_includedir}/*.h %{_libdir}/pkgconfig/* %{_mandir}/man3/* #%{_datadir}/aclocal/* %doc ACKNOWLEDGEMENTS ChangeLog NEWS NOTES README TODO +%files tools +%defattr(-,root,root) +%{_bindir}/sfconvert +%{_bindir}/sfinfo +%{_mandir}/man1/* + %changelog +* Sat Dec 12 2020 Silvan Calarco 0.3.6-2mamba +- rebuilt with debug package, -tools subpackage and patches from arch linux + * Fri Mar 08 2013 Automatic Build System 0.3.6-1mamba - automatic version update by autodist